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Network Working Group R. Braden
Request for Comments: 1633 ISI
Category: Informational D. Clark
MIT
S. Shenker
Xerox PARC
June 1994
Integrated Services in the Internet Architecture: an Overview
Status of this Memo
This memo provides information for the Internet community. This memo
does not specify an Internet standard of any kind. Distribution of
this memo is unlimited.
Abstract
This memo discusses a proposed extension to the Internet architecture
and protocols to provide integrated services, i.e., to support real-
time as well as the current non-real-time service of IP. This
extension is necessary to meet the growing need for real-time service
for a variety of new applications, including teleconferencing, remote
seminars, telescience, and distributed simulation.
This memo represents the direct product of recent work by Dave Clark,
Scott Shenker, Lixia Zhang, Deborah Estrin, Sugih Jamin, John
Wroclawski, Shai Herzog, and Bob Braden, and indirectly draws upon
the work of many others.
Table of Contents
1. Introduction ...................................................2
2. Elements of the Architecture ...................................3
2.1 Integrated Services Model ..................................3
2.2 Reference Implementation Framework .........................6
3. Integrated Services Model ......................................11
3.1 Quality of Service Requirements ............................12
3.2 Resource-Sharing Requirements and Service Models ...........16
3.3 Packet Dropping ............................................18
3.4 Usage Feedback .............................................19
3.5 Reservation Model ..........................................19
4. Traffic Control Mechanisms .....................................20
4.1 Basic Functions ............................................20
4.2 Applying the Mechanisms ....................................23
4.3 An example .................................................24
5. Reservation Setup Protocol .....................................25
5.1 RSVP Overview ..............................................25
5.2 Routing and Reservations ...................................28
6. Acknowledgments ................................................30
References ........................................................31
Security Considerations ...........................................32
Authors' Addresses ................................................33
1. Introduction
The multicasts of IETF meetings across the Internet have formed a
large-scale experiment in sending digitized voice and video through a
packet-switched infrastructure. These highly-visible experiments
have depended upon three enabling technologies. (1) Many modern
workstations now come equipped with built-in multimedia hardware,
including audio codecs and video frame-grabbers, and the necessary
video gear is now inexpensive. (2) IP multicasting, which is not yet
generally available in commercial routers, is being provided by the
MBONE, a temporary "multicast backbone". (3) Highly-sophisticated
digital audio and video applications have been developed.
These experiments also showed that an important technical element is
still missing: real-time applications often do not work well across
the Internet because of variable queueing delays and congestion
losses. The Internet, as originally conceived, offers only a very
simple quality of service (QoS), point-to-point best-effort data
delivery. Before real-time applications such as remote video,
multimedia conferencing, visualization, and virtual reality can be
broadly used, the Internet infrastructure must be modified to support
real-time QoS, which provides some control over end-to-end packet
delays. This extension must be designed from the beginning for
multicasting; simply generalizing from the unicast (point-to-point)
case does not work.
Real-time QoS is not the only issue for a next generation of traffic
management in the Internet. Network operators are requesting the
ability to control the sharing of bandwidth on a particular link
among different traffic classes. They want to be able to divide
traffic into a few administrative classes and assign to each a
minimum percentage of the link bandwidth under conditions of
overload, while allowing "unused" bandwidth to be available at other
times. These classes may represent different user groups or
different protocol families, for example. Such a management facility
is commonly called controlled link-sharing. We use the term
integrated services (IS) for an Internet service model that includes
best-effort service, real-time service, and controlled link sharing.
The requirements and mechanisms for integrated services have been the
subjects of much discussion and research over the past several years
(the literature is much too large to list even a representative
sample here; see the references in [CSZ92, Floyd92, Jacobson91,
JSCZ93, Partridge92, SCZ93, RSVP93a] for a partial list). This work
has led to the unified approach to integrated services support that
is described in this memo. We believe that it is now time to begin
the engineering that must precede deployment of integrated services
in the Internet.
Section 2 of this memo introduces the elements of an IS extension of
the Internet. Section 3 discusses real-time service models [SCZ93a,
SCZ93b]. Section 4 discusses traffic control, the forwarding
algorithms to be used in routers [CSZ92]. Section 5 discusses the
design of RSVP, a resource setup protocol compatible with the
assumptions of our IS model [RSVP93a, RSVP93b].
2. Elements of the Architecture
The fundamental service model of the Internet, as embodied in the
best-effort delivery service of IP, has been unchanged since the
beginning of the Internet research project 20 years ago [CerfKahn74].
We are now proposing to alter that model to encompass integrated
service. From an academic viewpoint, changing the service model of
the Internet is a major undertaking; however, its impact is mitigated
by the fact that we wish only to extend the original architecture.
The new components and mechanisms to be added will supplement but not
replace the basic IP service.
Abstractly, the proposed architectural extension is comprised of two
elements: (1) an extended service model, which we call the IS model,
and (2) a reference implementation framework, which gives us a set of
vocabulary and a generic program organization to realize the IS
model. It is important to separate the service model, which defines
the externally visible behavior, from the discussion of the
implementation, which may (and should) change during the life of the
service model. However, the two are related; to make the service
model credible, it is useful to provide an example of how it might be
realized.
2.1 Integrated Services Model
The IS model we are proposing includes two sorts of service
targeted towards real-time traffic: guaranteed and predictive
service. It integrates these services with controlled link-
sharing, and it is designed to work well with multicast as well as
unicast. Deferring a summary of the IS model to Section 3, we
first discuss some key assumptions behind the model.
The first assumption is that resources (e.g., bandwidth) must be
explicitly managed in order to meet application requirements.
This implies that "resource reservation" and "admission control"
are key building blocks of the service. An alternative approach,
which we reject, is to attempt to support real-time traffic
without any explicit changes to the Internet service model.
The essence of real-time service is the requirement for some
service guarantees, and we argue that guarantees cannot be
achieved without reservations. The term "guarantee" here is to be
broadly interpreted; they may be absolute or statistical, strict
or approximate. However, the user must be able to get a service
whose quality is sufficiently predictable that the application can
operate in an acceptable way over a duration of time determined by
the user. Again, "sufficiently" and "acceptable" are vague terms.
In general, stricter guarantees have a higher cost in resources
that are made unavailable for sharing with others.
The following arguments have been raised against resource
guarantees in the Internet.
o "Bandwidth will be infinite."
The incredibly large carrying capacity of an optical fiber
leads some to conclude that in the future bandwidth will be
so abundant, ubiquitous, and cheap that there will be no
communication delays other than the speed of light, and
therefore there will be no need to reserve resources.
However, we believe that this will be impossible in the short
term and unlikely in the medium term. While raw bandwidth
may seem inexpensive, bandwidth provided as a network service
is not likely to become so cheap that wasting it will be the
most cost-effective design principle. Even if low-cost
bandwidth does eventually become commonly available, we do
not accept that it will be available "everywhere" in the
Internet. Unless we provide for the possibility of dealing
with congested links, then real-time services will simply be
precluded in those cases. We find that restriction
unacceptable.
o "Simple priority is sufficient."
It is true that simply giving higher priority to real-time
traffic would lead to adequate real-time service at some
times and under some conditions. But priority is an
implementation mechanism, not a service model. If we define
the service by means of a specific mechanism, we may not get
the exact features we want. In the case of simple priority,
the issue is that as soon as there are too many real-time
streams competing for the higher priority, every stream is
degraded. Restricting our service to this single failure
mode is unacceptable. In some cases, users will demand that
some streams succeed while some new requests receive a "busy
signal".
o "Applications can adapt."
The development of adaptive real-time applications, such as
Jacobson's audio program VAT, does not eliminate the need to
bound packet delivery time. Human requirements for
interaction and intelligibility limit the possible range of
adaptation to network delays. We have seen in real
experiments that, while VAT can adapt to network delays of
many seconds, the users find that interaction is impossible
in these cases.
We conclude that there is an inescapable requirement for routers
to be able to reserve resources, in order to provide special QoS
for specific user packet streams, or "flows". This in turn
requires flow-specific state in the routers, which represents an
important and fundamental change to the Internet model. The
Internet architecture was been founded on the concept that all
flow-related state should be in the end systems [Clark88].
Designing the TCP/IP protocol suite on this concept led to a
robustness that is one of the keys to its success. In section 5
we discuss how the flow state added to the routers for resource
reservation can be made "soft", to preserve the robustness of the
Internet protocol suite.
There is a real-world side effect of resource reservation in
routers. Since it implies that some users are getting privileged
service, resource reservation will need enforcement of policy and
administrative controls. This in turn will lead to two kinds of
authentication requirements: authentication of users who make
reservation requests, and authentication of packets that use the
reserved resources. However, these issues are not unique to "IS";
other aspects of the evolution of the Internet, including
commercialization and commercial security, are leading to the same
requirements. We do not discuss the issues of policy or security
further in this memo, but they will require attention.
We make another fundamental assumption, that it is desirable to
use the Internet as a common infrastructure to support both non-
real-time and real-time communication. One could alternatively
build an entirely new, parallel infrastructure for real-time
services, leaving the Internet unchanged. We reject this
approach, as it would lose the significant advantages of
statistical sharing between real-time and non-real-time traffic,
and it would be much more complex to build and administer than a
common infrastructure.
In addition to this assumption of common infrastructure, we adopt
a unified protocol stack model, employing a single internet-layer
protocol for both real-time and non-real-time service. Thus, we
propose to use the existing internet-layer protocol (e.g., IP or
CLNP) for real-time data. Another approach would be to add a new
real-time protocol in the internet layer [ST2-90]. Our unified
stack approach provides economy of mechanism, and it allows us to
fold controlled link-sharing in easily. It also handles the
problem of partial coverage, i.e., allowing interoperation between
IS-capable Internet systems and systems that have not been
extended, without the complexity of tunneling.
We take the view that there should be a single service model for
the Internet. If there were different service models in different
parts of the Internet, it is very difficult to see how any end-
to-end service quality statements could be made. However, a
single service model does not necessarily imply a single
implementation for packet scheduling or admission control.
Although specific packet scheduling and admission control
mechanisms that satisfy our service model have been developed, it
is quite possible that other mechanisms will also satisfy the
service model. The reference implementation framework, introduced
below, is intended to allow discussion of implementation issues
without mandating a single design.
Based upon these considerations, we believe that an IS extension
that includes additional flow state in routers and an explicit
setup mechanism is necessary to provide the needed service. A
partial solution short of this point would not be a wise
investment. We believe that the extensions we propose preserve
the essential robustness and efficiency of the Internet
architecture, and they allow efficient management of the network
resources; these will be important goals even if bandwidth becomes
very inexpensive.
2.2 Reference Implementation Framework
We propose a reference implementation framework to realize the IS
model. This framework includes four components: the packet
scheduler, the admission control routine, the classifier, and the
reservation setup protocol. These are discussed briefly below and
more fully in Sections 4 and 5.
In the ensuing discussion, we define the "flow" abstraction as a
distinguishable stream of related datagrams that results from a
single user activity and requires the same QoS. For example, a
flow might consist of one transport connection or one video stream
between a given host pair. It is the finest granularity of packet
stream distinguishable by the IS. We define a flow to be simplex,
i.e., to have a single source but N destinations. Thus, an N-way
teleconference will generally require N flows, one originating at
each site.
In today's Internet, IP forwarding is completely egalitarian; all
packets receive the same quality of service, and packets are
typically forwarded using a strict FIFO queueing discipline. For
integrated services, a router must implement an appropriate QoS
for each flow, in accordance with the service model. The router
function that creates different qualities of service is called
"traffic control". Traffic control in turn is implemented by
three components: the packet scheduler, the classifier, and
admission control.
o Packet Scheduler
The packet scheduler manages the forwarding of different
packet streams using a set of queues and perhaps other
mechanisms like timers. The packet scheduler must be
implemented at the point where packets are queued; this is
the output driver level of a typical operating system, and
corresponds to the link layer protocol. The details of the
scheduling algorithm may be specific to the particular output
medium. For example, the output driver will need to invoke
the appropriate link-layer controls when interfacing to a
network technology that has an internal bandwidth allocation
mechanism.
An experimental packet scheduler has been built that
implements the IS model described in Section 3 and [SCZ93];
this is known as the CSZ scheduler and is discussed further
in Section 4. We note that the CSZ scheme is not mandatory
to accomplish our service model; indeed for parts of the
network that are known always to be underloaded, FIFO will
deliver satisfactory service.
There is another component that could be considered part of
the packet scheduler or separate: the estimator [Jacobson91].
This algorithm is used to measure properties of the outgoing
traffic stream, to develop statistics that control packet
scheduling and admission control. This memo will consider
the estimator to be a part of the packet scheduler.
o Classifier
For the purpose of traffic control (and accounting), each
incoming packet must be mapped into some class; all packets
in the same class get the same treatment from the packet
scheduler. This mapping is performed by the classifier.
Choice of a class may be based upon the contents of the
existing packet header(s) and/or some additional
classification number added to each packet.
A class might correspond to a broad category of flows, e.g.,
all video flows or all flows attributable to a particular
organization. On the other hand, a class might hold only a
single flow. A class is an abstraction that may be local to
a particular router; the same packet may be classified
differently by different routers along the path. For
example, backbone routers may choose to map many flows into a
few aggregated classes, while routers nearer the periphery,
where there is much less aggregation, may use a separate
class for each flow.
o Admission Control
Admission control implements the decision algorithm that a
router or host uses to determine whether a new flow can be
granted the requested QoS without impacting earlier
guarantees. Admission control is invoked at each node to
make a local accept/reject decision, at the time a host
requests a real-time service along some path through the
Internet. The admission control algorithm must be consistent
with the service model, and it is logically part of traffic
control. Although there are still open research issues in
admission control, a first cut exists [JCSZ92].
Admission control is sometimes confused with policing or
enforcement, which is a packet-by-packet function at the
"edge" of the network to ensure that a host does not violate
its promised traffic characteristics. We consider policing
to be one of the functions of the packet scheduler.
In addition to ensuring that QoS guarantees are met,
admission control will be concerned with enforcing
administrative policies on resource reservations. Some
policies will demand authentication of those requesting
reservations. Finally, admission control will play an
important role in accounting and administrative reporting.
The fourth and final component of our implementation framework is
a reservation setup protocol, which is necessary to create and
maintain flow-specific state in the endpoint hosts and in routers
along the path of a flow. Section discusses a reservation setup
protocol called RSVP (for "ReSerVation Protocol") [RSVP93a,
RSVP93b]. It may not be possible to insist that there be only one
reservation protocol in the Internet, but we will argue that
multiple choices for reservation protocols will cause confusion.
We believe that multiple protocols should exist only if they
support different modes of reservation.
The setup requirements for the link-sharing portion of the service
model are far less clear than those for resource reservations.
While we expect that much of this can be done through network
management interfaces, and thus need not be part of the overall
architecture, we may also need RSVP to play a role in providing
the required state.
In order to state its resource requirements, an application must
specify the desired QoS using a list of parameters that is called
a "flowspec" [Partridge92]. The flowspec is carried by the
reservation setup protocol, passed to admission control for to
test for acceptability, and ultimately used to parametrize the
packet scheduling mechanism.
Figure shows how these components might fit into an IP router
that has been extended to provide integrated services. The router
has two broad functional divisions: the forwarding path below the
double horizontal line, and the background code above the line.
The forwarding path of the router is executed for every packet and
must therefore be highly optimized. Indeed, in most commercial
routers, its implementation involves a hardware assist. The
forwarding path is divided into three sections: input driver,
internet forwarder, and output driver. The internet forwarder
interprets the internetworking protocol header appropriate to the
protocol suite, e.g., the IP header for TCP/IP, or the CLNP header
for OSI. For each packet, an internet forwarder executes a
suite-dependent classifier and then passes the packet and its
class to the appropriate output driver. A classifier must be both
general and efficient. For efficiency, a common mechanism should
be used for both resource classification and route lookup.
The output driver implements the packet scheduler. (Layerists
will observe that the output driver now has two distinct sections:
the packet scheduler that is largely independent of the detailed
mechanics of the interface, and the actual I/O driver that is only
concerned with the grittiness of the hardware. The estimator
lives somewhere in between. We only note this fact, without
suggesting that it be elevated to a principle.).
_____________________________________________________________
| ____________ ____________ ___________ |
| | | | Reservation| | | |
| | Routing | | Setup | | Management| |
| | Agent | | Agent | | Agent | |
| |______._____| |______._____| |_____._____| |
| . . | . |
| . . _V________ . |
| . . | Admission| . |
| . . | Control | . |
| V . |__________| . |
| [Routing ] V V |
| [Database] [Traffic Control Database] |
|=============================================================|
| | | _______ |
| | __________ | |_|_|_|_| => o |
| | | | | Packet | _____ |
| ====> |Classifier| =====> Scheduler |===>|_|_|_| ===>
| | |__________| | _______ | |
| | | |_|_|_|_| => o |
| Input | Internet | |
| Driver | Forwarder | O u t p u t D r i v e r |
|________|__________________|_________________________________|
Figure 1: Implementation Reference Model for Routers
The background code is simply loaded into router memory and
executed by a general-purpose CPU. These background routines
create data structures that control the forwarding path. The
routing agent implements a particular routing protocol and builds
a routing database. The reservation setup agent implements the
protocol used to set up resource reservations; see Section . If
admission control gives the "OK" for a new request, the
appropriate changes are made to the classifier and packet
scheduler database to implement the desired QoS. Finally, every
router supports an agent for network management. This agent must
be able to modify the classifier and packet scheduler databases to
set up controlled link-sharing and to set admission control
policies.
The implementation framework for a host is generally similar to
that for a router, with the addition of applications. Rather than
being forwarded, host data originates and terminates in an
application. An application needing a real-time QoS for a flow
must somehow invoke a local reservation setup agent. The best way
to interface to applications is still to be determined. For
example, there might be an explicit API for network resource
setup, or the setup might be invoked implicitly as part of the
operating system scheduling function. The IP output routine of a
host may need no classifier, since the class assignment for a
packet can be specified in the local I/O control structure
corresponding to the flow.
In routers, integrated service will require changes to both the
forwarding path and the background functions. The forwarding
path, which may depend upon hardware acceleration for performance,
will be the more difficult and costly to change. It will be vital
to choose a set of traffic control mechanisms that is general and
adaptable to a wide variety of policy requirements and future
circumstances, and that can be implemented efficiently.
3. Integrated Services Model
A service model is embedded within the network service interface
invoked by applications to define the set of services they can
request. While both the underlying network technology and the
overlying suite of applications will evolve, the need for
compatibility requires that this service interface remain relatively
stable (or, more properly, extensible; we do expect to add new
services in the future but we also expect that it will be hard to
change existing services). Because of its enduring impact, the
service model should not be designed in reference to any specific
network artifact but rather should be based on fundamental service
requirements.
We now briefly describe a proposal for a core set of services for the
Internet; this proposed core service model is more fully described in
[SCZ93a, SCZ93b]. This core service model addresses those services
which relate most directly to the time-of-delivery of packets. We
leave the remaining services (such as routing, security, or stream
synchronization) for other standardization venues. A service model
consists of a set of service commitments; in response to a service
request the network commits to deliver some service. These service
commitments can be categorized by the entity to whom they are made:
they can be made to either individual flows or to collective entities
(classes of flows). The service commitments made to individual flows
are intended to provide reasonable application performance, and thus
are driven by the ergonomic requirements of the applications; these
service commitments relate to the quality of service delivered to an
individual flow. The service commitments made to collective entities
are driven by resource-sharing, or economic, requirements; these
service commitments relate to the aggregate resources made available
to the various entities.
In this section we start by exploring the service requirements of
individual flows and propose a corresponding set of services. We
then discuss the service requirements and services for resource
sharing. Finally, we conclude with some remarks about packet
dropping.
3.1 Quality of Service Requirements
The core service model is concerned almost exclusively with the
time-of-delivery of packets. Thus, per-packet delay is the
central quantity about which the network makes quality of service
commitments. We make the even more restrictive assumption that
the only quantity about which we make quantitative service
commitments are bounds on the maximum and minimum delays.
The degree to which application performance depends on low delay
service varies widely, and we can make several qualitative
distinctions between applications based on the degree of their
dependence. One class of applications needs the data in each
packet by a certain time and, if the data has not arrived by then,
the data is essentially worthless; we call these real-time
applications. Another class of applications will always wait for
data to arrive; we call these " elastic" applications. We now
consider the delay requirements of these two classes separately.
3.1.1 Real-Time Applications
An important class of such real-time applications, which are
the only real-time applications we explicitly consider in the
arguments that follow, are "playback" applications. In a
playback application, the source takes some signal, packetizes
it, and then transmits the packets over the network. The
network inevitably introduces some variation in the delay of
the delivered packets. The receiver depacketizes the data and
then attempts to faithfully play back the signal. This is done
by buffering the incoming data and then replaying the signal at
some fixed offset delay from the original departure time; the
term "playback point" refers to the point in time which is
offset from the original departure time by this fixed delay.
Any data that arrives before its associated playback point can
be used to reconstruct the signal; data arriving after the
playback point is essentially useless in reconstructing the
real-time signal.
In order to choose a reasonable value for the offset delay, an
application needs some "a priori" characterization of the
maximum delay its packets will experience. This "a priori"
characterization could either be provided by the network in a
quantitative service commitment to a delay bound, or through
the observation of the delays experienced by the previously
arrived packets; the application needs to know what delays to
expect, but this expectation need not be constant for the
entire duration of the flow.
The performance of a playback application is measured along two
dimensions: latency and fidelity. Some playback applications,
in particular those that involve interaction between the two
ends of a connection such as a phone call, are rather sensitive
to the latency; other playback applications, such as
transmitting a movie or lecture, are not. Similarly,
applications exhibit a wide range of sensitivity to loss of
fidelity. We will consider two somewhat artificially
dichotomous classes: intolerant applications, which require an
absolutely faithful playback, and tolerant applications, which
can tolerate some loss of fidelity. We expect that the vast
bulk of audio and video applications will be tolerant, but we
also suspect that there will be other applications, such as
circuit emulation, that are intolerant.
Delay can affect the performance of playback applications in
two ways. First, the value of the offset delay, which is
determined by predictions about the future packet delays,
determines the latency of the application. Second, the delays
of individual packets can decrease the fidelity of the playback
by exceeding the offset delay; the application then can either
change the offset delay in order to play back late packets
(which introduces distortion) or merely discard late packets
(which creates an incomplete signal). The two different ways
of coping with late packets offer a choice between an
incomplete signal and a distorted one, and the optimal choice
will depend on the details of the application, but the
important point is that late packets necessarily decrease
fidelity.
Intolerant applications must use a fixed offset delay, since
any variation in the offset delay will introduce some
distortion in the playback. For a given distribution of packet
delays, this fixed offset delay must be larger than the
absolute maximum delay, to avoid the possibility of late
packets. Such an application can only set its offset delay
appropriately if it is given a perfectly reliable upper bound
on the maximum delay of each packet. We call a service
characterized by a perfectly reliable upper bound on delay "
guaranteed service", and propose this as the appropriate
service model for intolerant playback applications.
In contrast, tolerant applications need not set their offset
delay greater than the absolute maximum delay, since they can
tolerate some late packets. Moreover, instead of using a
single fixed value for the offset delay, they can attempt to
reduce their latency by varying their offset delays in response
to the actual packet delays experienced in the recent past. We
call applications which vary their offset delays in this manner
"adaptive" playback applications.
For tolerant applications we propose a service model called "
predictive service" which supplies a fairly reliable, but not
perfectly reliable, delay bound. This bound, in contrast to
the bound in the guaranteed service, is not based on worst case
assumptions on the behavior of other flows. Instead, this
bound might be computed with properly conservative predictions
about the behavior of other flows. If the network turns out to
be wrong and the bound is violated, the application's
performance will perhaps suffer, but the users are willing to
tolerate such interruptions in service in return for the
presumed lower cost of the service. Furthermore, because many
of the tolerant applications are adaptive, we augment the
predictive service to also give "minimax" service, which is to
attempt to minimize the ex post maximum delay. This service is
not trying to minimize the delay of every packet, but rather is
trying to pull in the tail of the delay distribution.
It is clear that given a choice, with all other things being
equal, an application would perform no worse with absolutely
reliable bounds than with fairly reliable bounds. Why, then,
do we offer predictive service? The key consideration here is
efficiency; when one relaxes the service requirements from
perfectly to fairly reliable bounds, this increases the level
of network utilization that can be sustained, and thus the
price of the predictive service will presumably be lower than
that of guaranteed service. The predictive service class is
motivated by the conjecture that the performance penalty will
be small for tolerant applications but the overall efficiency
gain will be quite large.
In order to provide a delay bound, the nature of the traffic
from the source must be characterized, and there must be some
admission control algorithm which insures that a requested flow
can actually be accommodated. A fundamental point of our
overall architecture is that traffic characterization and
admission control are necessary for these real-time delay bound
services. So far we have assumed that an application's data
generation process is an intrinsic property unaffected by the
network. However, there are likely to be many audio and video
applications which can adjust their coding scheme and thus can
alter the resulting data generation process depending on the
network service available. This alteration of the coding
scheme will present a tradeoff between fidelity (of the coding
scheme itself, not of the playback process) and the bandwidth
requirements of the flow. Such "rate-adaptive" playback
applications have the advantage that they can adjust to the
current network conditions not just by resetting their playback
point but also by adjusting the traffic pattern itself. For
rate-adaptive applications, the traffic characterizations used
in the service commitment are not immutable. We can thus
augment the service model by allowing the network to notify
(either implicitly through packet drops or explicitly through
control packets) rate-adaptive applications to change their
traffic characterization.
3.1.2 Elastic Applications
While real-time applications do not wait for late data to
arrive, elastic applications will always wait for data to
arrive. It is not that these applications are insensitive to
delay; to the contrary, significantly increasing the delay of a
packet will often harm the application's performance. Rather,
the key point is that the application typically uses the
arriving data immediately, rather than buffering it for some
later time, and will always choose to wait for the incoming
data rather than proceed without it. Because arriving data can
be used immediately, these applications do not require any a
priori characterization of the service in order for the
application to function. Generally speaking, it is likely that
for a given distribution of packet delays, the perceived
performance of elastic applications will depend more on the
average delay than on the tail of the delay distribution. One
can think of several categories of such elastic applications:
interactive burst (Telnet, X, NFS), interactive bulk transfer
(FTP), and asynchronous bulk transfer (electronic mail, FAX).
The delay requirements of these elastic applications vary from
rather demanding for interactive burst applications to rather
lax for asynchronous bulk transfer, with interactive bulk
transfer being intermediate between them.
An appropriate service model for elastic applications is to
provide "as-soon-as-possible", or ASAP service. (For
compatibility with historical usage, we will use the term
best-effort service when referring to ASAP service.). We
furthermore propose to offer several classes of best-effort
service to reflect the relative delay sensitivities of
different elastic applications. This service model allows
interactive burst applications to have lower delays than
interactive bulk applications, which in turn would have lower
delays than asynchronous bulk applications. In contrast to the
real-time service models, applications using this service are
not subject to admission control.
The taxonomy of applications into tolerant playback, intolerant
playback, and elastic is neither exact nor complete, but was
only used to guide the development of the core service model.
The resulting core service model should be judged not on the
validity of the underlying taxonomy but rather on its ability
to adequately meet the needs of the entire spectrum of
applications. In particular, not all real-time applications
are playback applications; for example, one might imagine a
visualization application which merely displayed the image
encoded in each packet whenever it arrived. However, non-
playback applications can still use either the guaranteed or
predictive real-time service model, although these services are
not specifically tailored to their needs. Similarly, playback
applications cannot be neatly classified as either tolerant or
intolerant, but rather fall along a continuum; offering both
guaranteed and predictive service allows applications to make
their own tradeoff between fidelity, latency, and cost.
Despite these obvious deficiencies in the taxonomy, we expect
that it describes the service requirements of current and
future applications well enough so that our core service model
can adequately meet all application needs.
3.2 Resource-Sharing Requirements and Service Models
The last section considered quality of service commitments; these
commitments dictate how the network must allocate its resources
among the individual flows. This allocation of resources is
typically negotiated on a flow-by-flow basis as each flow requests
admission to the network, and does not address any of the policy
issues that arise when one looks at collections of flows. To
address these collective policy issues, we now discuss resource-
sharing service commitments. Recall that for individual quality
of service commitments we focused on delay as the only quantity of
interest. Here, we postulate that the quantity of primary
interest in resource-sharing is aggregate bandwidth on individual
links. Thus, this component of the service model, called "link-
sharing", addresses the question of how to share the aggregate
bandwidth of a link among various collective entities according to
some set of specified shares. There are several examples that are
commonly used to explain the requirement of link-sharing among
collective entities.
Multi-entity link-sharing. -- A link may be purchased and used
jointly by several organizations, government agencies or the like.
They may wish to insure that under overload the link is shared in
a controlled way, perhaps in proportion to the capital investment
of each entity. At the same time, they might wish that when the
link is underloaded, any one of the entities could utilize all the
idle bandwidth.
Multi-protocol link-sharing -- In a multi-protocol Internet, it
may be desired to prevent one protocol family (DECnet, IP, IPX,
OSI, SNA, etc.) from overloading the link and excluding the other
families. This is important because different families may have
different methods of detecting and responding to congestion, and
some methods may be more "aggressive" than others. This could lead
to a situation in which one protocol backs off more rapidly than
another under congestion, and ends up getting no bandwidth.
Explicit control in the router may be required to correct this.
Again, one might expect that this control should apply only under
overload, while permitting an idle link to be used in any
proportion.
Multi-service sharing -- Within a protocol family such as IP, an
administrator might wish to limit the fraction of bandwidth
allocated to various service classes. For example, an
administrator might wish to limit the amount of real-time traffic
to some fraction of the link, to avoid preempting elastic traffic
such as FTP.
In general terms, the link-sharing service model is to share the
aggregate bandwidth according to some specified shares. We can
extend this link-sharing service model to a hierarchical version.
For instance, a link could be divided between a number of
organizations, each of which would divide the resulting allocation
among a number of protocols, each of which would be divided among
a number of services. Here, the sharing is defined by a tree with
shares assigned to each leaf node.
An idealized fluid model of instantaneous link-sharing with
proportional sharing of excess is the fluid processor sharing
model (introduced in [DKS89] and further explored in [Parekh92]
and generalized to the hierarchical case) where at every instant
the available bandwidth is shared between the active entities
(i.e., those having packets in the queue) in proportion to the
assigned shares of the resource. This fluid model exhibits the
desired policy behavior but is, of course, an unrealistic
idealization. We then propose that the actual service model
should be to approximate, as closely as possible, the bandwidth
shares produced by this ideal fluid model. It is not necessary to
require that the specific order of packet departures match those
of the fluid model since we presume that all detailed per-packet
delay requirements of individual flows are addressed through
quality of service commitments and, furthermore, the satisfaction
with the link-sharing service delivered will probably not depend
very sensitively on small deviations from the scheduling implied
by the fluid link-sharing model.
We previously observed that admission control was necessary to
ensure that the real-time service commitments could be met.
Similarly, admission control will again be necessary to ensure
that the link-sharing commitments can be met. For each entity,
admission control must keep the cumulative guaranteed and
predictive traffic from exceeding the assigned link-share.
3.3 Packet Dropping
So far, we have implicitly assumed that all packets within a flow
were equally important. However, in many audio and video streams,
some packets are more valuable than others. We therefore propose
augmenting the service model with a "preemptable" packet service,
whereby some of the packets within a flow could be marked as
preemptable. When the network was in danger of not meeting some
of its quantitative service commitments, it could exercise a
certain packet's "preemptability option" and discard the packet
(not merely delay it, since that would introduce out-of-order
problems). By discarding these preemptable packets, a router can
reduce the delays of the not-preempted packets.
Furthermore, one can define a class of packets that is not subject
to admission control. In the scenario described above where
preemptable packets are dropped only when quantitative service
commitments are in danger of being violated, the expectation is
that preemptable packets will almost always be delivered and thus
they must included in the traffic description used in admission
control. However, we can extend preemptability to the extreme
case of "expendable" packets (the term expendable is used to
connote an extreme degree of preemptability), where the
expectation is that many of these expendable packets may not be
delivered. One can then exclude expendable packets from the
traffic description used in admission control; i.e., the packets
are not considered part of the flow from the perspective of
admission control, since there is no commitment that they will be
delivered.
3.4 Usage Feedback
Another important issue in the service is the model for usage
feedback, also known as "accounting", to prevent abuse of network
resources. The link-sharing service described earlier can be
used to provide administratively-imposed limits on usage.
However, a more free-market model of network access will require
back-pressure on users for the network resources they reserve.
This is a highly contentious issue, and we are not prepared to say
more about it at this time.
3.5 Reservation Model
The "reservation model" describes how an application negotiates
for a QoS level. The simplest model is that the application asks
for a particular QoS and the network either grants it or refuses.
Often the situation will be more complex. Many applications will
be able to get acceptable service from a range of QoS levels, or
more generally, from anywhere within some region of the multi-
dimensional space of a flowspec.
For example, rather than simply refusing the request, the network
might grant a lower resource level and inform the application of
what QoS has been actually granted. A more complex example is the
"two-pass" reservation model, In this scheme, an "offered"
flowspec is propagated along the multicast distribution tree from
each sender Si to all receivers Rj. Each router along the path
records these values and perhaps adjusts them to reflect available
capacity. The receivers get these offers, generate corresponding
"requested" flowspecs, and propagate them back along the same
routes to the senders. At each node, a local reconciliation must
be performed between the offered and the requested flowspec to
create a reservation, and an appropriately modified requested
flowspec is passed on. This two-pass scheme allows extensive
properties like allowed delay to be distributed across hops in the
path [Tenet90, ST2-90]. Further work is needed to define the
amount of generality, with a corresponding level of complexity,
that is required in the reservation model.
4. Traffic Control Mechanisms
We first survey very briefly the possible traffic control mechanisms.
Then in Section 4.2 we apply a subset of these mechanisms to support
the various services that we have proposed.
4.1 Basic Functions
In the packet forwarding path, there is actually a very limited
set of actions that a router can take. Given a particular packet,
a router must select a route for it; in addition the router can
either forward it or drop it, and the router may reorder it with
respect to other packets waiting to depart. The router can also
hold the packet, even though the link is idle. These are the
building blocks from which we must fashion the desired behavior.
4.1.1 Packet Scheduling
The basic function of packet scheduling is to reorder the
output queue. There are many papers that have been written on
possible ways to manage the output queue, and the resulting
behavior. Perhaps the simplest approach is a priority scheme,
in which packets are ordered by priority, and highest priority
packets always leave first. This has the effect of giving some
packets absolute preference over others; if there are enough of
the higher priority packets, the lower priority class can be
completely prevented from being sent.
An alternative scheduling scheme is round-robin or some
variant, which gives different classes of packets access to a
share of the link. A variant called Weighted Fair Queueing, or
WFQ, has been demonstrated to allocate the total bandwidth of a
link into specified shares.
There are more complex schemes for queue management, most of
which involve observing the service objectives of individual
packets, such as delivery deadline, and ordering packets based
on these criteria.
4.1.2 Packet Dropping
The controlled dropping of packets is as important as their
scheduling.
Most obviously, a router must drop packets when its buffers are
all full. This fact, however, does not determine which packet
should be dropped. Dropping the arriving packet, while simple,
may cause undesired behavior.
In the context of today's Internet, with TCP operating over
best effort IP service, dropping a packet is taken by TCP as a
signal of congestion and causes it to reduce its load on the
network. Thus, picking a packet to drop is the same as picking
a source to throttle. Without going into any particular
algorithm, this simple relation suggests that some specific
dropping controls should be implemented in routers to improve
congestion control.
In the context of real-time services, dropping more directly
relates to achieving the desired quality of service. If a
queue builds up, dropping one packet reduces the delay of all
the packets behind it in the queue. The loss of one can
contribute to the success of many. The problem for the
implementor is to determine when the service objective (the
delay bound) is in danger of being violated. One cannot look
at queue length as an indication of how long packets have sat
in a queue. If there is a priority scheme in place, packets of
lower priority can be pre-empted indefinitely, so even a short
queue may have very old packets in it. While actual time
stamps could be used to measure holding time, the complexity
may be unacceptable.
Some simple dropping schemes, such as combining all the buffers
in a single global pool, and dropping the arriving packet if
the pool is full, can defeat the service objective of a WFQ
scheduling scheme. Thus, dropping and scheduling must be
coordinated.
4.1.3 Packet Classification
The above discussion of scheduling and dropping presumed that
the packet had been classified into some flow or sequence of
packets that should be treated in a specified way. A
preliminary to this sort of processing is the classification
itself. Today a router looks at the destination address and
selects a route. The destination address is not sufficient to
select the class of service a packet must receive; more
information is needed.
One approach would be to abandon the IP datagram model for a
virtual circuit model, in which a circuit is set up with
specific service attributes, and the packet carries a circuit
identifier. This is the approach of ATM as well as protocols
such as ST-II [ST2-90]. Another model, less hostile to IP, is
to allow the classifier to look at more fields in the packet,
such as the source address, the protocol number and the port
fields. Thus, video streams might be recognized by a
particular well-known port field in the UDP header, or a
particular flow might be recognized by looking at both the
source and destination port numbers. It would be possible to
look even deeper into the packets, for example testing a field
in the application layer to select a subset of a
hierarchically-encoded video stream.
The classifier implementation issues are complexity and
processing overhead. Current experience suggests that careful
implementation of efficient algorithms can lead to efficient
classification of IP packets. This result is very important,
since it allows us to add QoS support to existing applications,
such as Telnet, which are based on existing IP headers.
One approach to reducing the overhead of classification would
be to provide a "flow-id" field in the Internet-layer packet
header. This flow-id would be a handle that could be cached
and used to short-cut classification of the packet. There are
a number of variations of this concept, and engineering is
required to choose the best design.
4.1.4 Admission Control
As we stated in the introduction, real-time service depends on
setting up state in the router and making commitments to
certain classes of packets. In order to insure that these
commitments can be met, it is necessary that resources be
explicitly requested, so that the request can be refused if the
resources are not available. The decision about resource
availability is called admission control.
Admission control requires that the router understand the
demands that are currently being made on its assets. The
approach traditionally proposed is to remember the service
parameters of past requests, and make a computation based on
the worst-case bounds on each service. A recent proposal,
which is likely to provide better link utilization, is to
program the router to measure the actual usage by existing
packet flows, and to use this measured information as a basis
of admitting new flows [JCSZ92]. This approach is subject to
higher risk of overload, but may prove much more effective in
using bandwidth.
Note that while the need for admission control is part of the
global service model, the details of the algorithm run in each
router is a local matter. Thus, vendors can compete by
developing and marketing better admission control algorithms,
which lead to higher link loadings with fewer service
overloads.
4.2 Applying the Mechanisms
The various tools described above can be combined to support the
services which were discussed in section 3.
o Guaranteed delay bounds
A theoretical result by Parekh [Parekh92] shows that if the
router implements a WFQ scheduling discipline, and if the
nature of the traffic source can be characterized (e.g. if it
fits within some bound such as a token bucket) then there
will be an absolute upper bound on the network delay of the
traffic in question. This simple and very powerful result
applies not just to one switch, but to general networks of
routers. The result is a constructive one; that is, Parekh
displays a source behavior which leads to the bound, and then
shows that this behavior is the worst possible. This means
that the bound he computes is the best there can be, under
these assumptions.
o Link sharing
The same WFQ scheme can provide controlled link sharing. The
service objective here is not to bound delay, but to limit
overload shares on a link, while allowing any mix of traffic
to proceed if there is spare capacity. This use of WFQ is
available in commercial routers today, and is used to
segregate traffic into classes based on such things as
protocol type or application. For example, one can allocate
separate shares to TCP, IPX and SNA, and one can assure that
network control traffic gets a guaranteed share of the link.
o Predictive real-time service
This service is actually more subtle than guaranteed service.
Its objective is to give a delay bound which is, on the one
hand, as low as possible, and on the other hand, stable
enough that the receiver can estimate it. The WFQ mechanism
leads to a guaranteed bound, but not necessarily a low bound.
In fact, mixing traffic into one queue, rather than
separating it as in WFQ, leads to lower bounds, so long as
the mixed traffic is generally similar (e.g., mixing traffic
from multiple video coders makes sense, mixing video and FTP
does not).
This suggests that we need a two-tier mechanism, in which the
first tier separates traffic which has different service
objectives, and the second tier schedules traffic within each
first tier class in order to meet its service objective.
4.3 An example: The CSZ scheme
As a proof of concept, a code package has been implemented which
realizes the services discussed above. It actually uses a number
of the basic tools, combined in a way specific to the service
needs. We describe in general terms how it works, to suggest how
services can be realized. We stress that there are other ways of
building a router to meet the same service needs, and there are in
fact other implementations being used today.
At the top level, the CSZ code uses WFQ as an isolation mechanism
to separate guaranteed flows from each other, as well as from the
rest of the traffic. Guaranteed service gets the highest priority
when and only when it needs the access to meets its deadline. WFQ
provides a separate guarantee for each and every guaranteed flow.
Predictive service and best effort service are separated by
priority. Within the predictive service class, a further priority
is used to provide sub-classes with different delay bounds.
Inside each predictive sub-class, simple FIFO queueing is used to
mix the traffic, which seems to produce good overall delay
behavior. This works because the top-tier algorithm has separated
out the best effort traffic such as FTP.
Within the best-effort class, WFQ is used to provide link sharing.
Since there is a possible requirement for nested shares, this WFQ
code can be used recursively. There are thus two different uses
of WFQ in this code, one to segregate the guaranteed classes, and
one to segregate the link shares. They are similar, but differ in
detail.
Within each link share of the best effort class, priority is used
to permit more time-sensitive elastic traffic to precede other
elastic traffic, e.g., to allow interactive traffic to precede
asynchronous bulk transfers.
The CSZ code thus uses both WFQ and priority in an alternating
manner to build a mechanism to support a range of rather
sophisticated service offerings. This discussion is very brief,
and does not touch on a number of significant issues, such as how
the CSZ code fits real time traffic into the link sharing
objectives. But the basic building blocks are very simple, and
very powerful. In particular, while priority has been proposed as
a key to real-time services, WFQ may be the more general and
powerful of the two schemes. It, rather than priority, supports
guaranteed service and link sharing.
5. Reservation Setup Protocol
There are a number of requirements to be met by the design of a
reservation setuop protocol. It should be fundamentally designed for
a multicast environment, and it must accommodate heterogeneous
service needs. It must give flexible control over the manner in
which reservations can be shared along branches of the multicast
delivery trees. It should be designed around the elementary action
of adding one sender and/or receiver to an existing set, or deleting
one. It must be robust and scale well to large multicast groups.
Finally, it must provide for advance reservation of resources, and
for the preemption that this implies. The reservation setup protocol
RSVP has been designed to meet these requirements [RSVP93a, RSVP93b].
This section gives an overview of the design of RSVP.
5.1 RSVP Overview
Figure shows multi-source, multi-destination data delivery for a
particular shared, distributed application. The arrows indicate
data flow from senders S1 and S2 to receivers R1, R2, and R3, and
the cloud represents the distribution mesh created by the
multicast routing protocol. Multicasting distribution replicates
each data packet from a sender Si, for delivery to every receiver
Rj. We treat uncast delivery from S1 to R1 as a special case, and
we call this multicast distribution mesh a session. A session is
defined by the common IP (multicast) destination address of the
receiver(s).
Senders Receivers
_____________________
( ) ===> R1
S1 ===> ( Multicast )
( ) ===> R2
( distribution )
S2 ===> ( )
( ) ===> R3
(_____________________)
Figure 2: Multicast Distribution Session
5.1.1 Flowspecs and Filter Specs
In general, an RSVP reservation request specifies the amount of
resources to be reserved for all, or some subset of, the
packets in a particular session. The resource quantity is
specified by a flowspec, while the packet subset to receive
those resources is specified by a filter spec. Assuming
admission control succeeds, the flowspec will be used to
parametrize a resource class in the packet scheduler, and the
filter spec will be instantiated in the packet classifier to
map the appropriate packets into this class. The subset of the
classifier state that selects a particular class is referred to
in RSVP documentation as a (packet) "filter".
The RSVP protocol mechanisms provide a very general facility
for creating and maintaining distributed reservation state
across the mesh of multicast delivery paths. These mechanisms
treat flowspecs and filter specs as mostly opaque binary data,
handing them to the local traffic control machinery for
interpretation. Of course, the service model presented to an
application must specify how to encode flowspecs and filter
specs.
5.1.2 Reservation Styles
RSVP offers several different reservation "styles", which
determine the manner in which the resource requirements of
multiple receivers are aggregated in the routers. These styles
allow the reserved resources to more efficiently meet
application requirements. Currently there are three
reservation styles, "wildcard", "fixed-filter", and " dynamic-
filter". A wildcard reservation uses a filter spec that is not
source-specific, so all packets destined for the associated
destination (session) may use a common pool of reserved
resources. This allows a single resource allocation to be made
across all distribution paths for the group. The wildcard
reservation style is useful in support of an audio conference,
where at most a small number of sources are active
simultaneously and may share the resource allocation.
The other two styles use filter specs that select particular
sources. A receiver may desire to receive from a fixed set of
sources, or instead it may desire the network to switch between
different source, by changing its filter spec(s) dymamically.
A fixed-filter style reservation cannot be changed during its
lifetime without re-invoking admission control. Dynamic-filter
reservations do allow a receiver to modify its choice of
source(s) over time without additional admission control;
however, this requires that sufficient resources be allocated
to handle the worst case when all downstream receivers take
input from different sources.
5.1.3 Receiver Initiation
An important design question is whether senders or receivers
should have responsibility for initiating reservations. A
sender knows the qualities of the traffic stream it can send,
while a receiver knows what it wants to (or can) receive.
Perhaps the most obvious choice is to let the sender initiate
the reservation. However, this scales poorly for large,
dynamic multicast delivery trees and for heterogeneous
receivers.
Both of these scaling problems are solved by making the
receiver responsible for initiating a reservation. Receiver
initiation handles heterogeneous receivers easily; each
receiver simply asks for a reservation appropriate to itself,
and any differences among reservations from different receivers
are resolved ("merged") within the network by RSVP. Receiver
initiation is also consisent with IP multicast, in which a
multicast group is created implicitly by receivers joining it.
Although receiver-initiated reservation is the natural choice
for multicast sessions, the justification for receiver
initiateion may appear weaker for unicast sessions, where the
sender may be the logical session initiator. However, we
expect that every realtime application will have its higher-
level signalling and control protocol, and this protocol can be
used to signal the receiver to initiate a reservation (and
perhaps indicate the flowspec to be used). For simplicity and
economy, a setup protocol should support only one direction of
initiation, and, and receiver initiation appears to us to be
the clear winner.
RSVP uses receiver-initiation of rservations [RSVP93b]. A
receiver is assumed to learn the senders' offered flowspecs by
a higher-level mechanism ("out of band"), it then generates its
own desired flowspec and propagates it towards the senders,
making reservations in each router along the way.
5.1.4 Soft State
There are two different possible styles for reservation setup
protocols, the "hard state" (HS) approach (also called
"connection-oriented"), and the "soft state" (SS) approach
(also called "connectionless"). In both approaches, multicast
distribution is performed using flow-specific state in each
router along the path. Under the HS approach, this state is
created and deleted in a fully deterministic manner by
cooperation among the routers. Once a host requests a session,
the "network" takes responsibility for creating and later
destroying the necessary state. ST-II is an example of the HS
approach [ST2-90]. Since management of HS session state is
completely deterministic, the HS setup protocol must be
reliable, with acknowledgments and retransmissions. In order
to achieve deterministic cleanup of state after a failure,
there must be some mechanism to detect failures, i.e., an
"up/down" protocol. The router upstream (towards the source)
from a failure takes responsibility for rebuilding the
necessary state on the router(s) along an alternate route.
RSVP takes the SS approach, which regards the reservation state
as cached information that is installed and periodically
refreshed by the end hosts. Unused state is timed out by the
routers. If the route changes, the refresh messages
automatically install the necessary state along the new route.
The SS approach was chosen to obtain the simplicity and
robustness that have been demonstrated by connectionless
protocols such as IP [Clark88].
5.2 Routing and Reservations
There is a fundamental interaction between resource reservation
set up and routing, since reservation requires the installation of
flow state along the route of data packets. If and when a route
changes, there must be some mechanism to set up a reservation
along the new route.
Some have suggested that reservation setup necessarily requires
route set up, i.e., the imposition of a virtual-circuit internet
layer. However, our goal is to simply extend the Internet
architecture, not replace it. The fundamental connectionless
internet layer [Clark88] has been highly successful, and we wish
to retain it as an architectural foundation. We propose instead
to modify somewhat the pure datagram forwarding mechanism of the
present Internet to accomodate "IS".
There are four routing issues faced by a reservation setup
protocol such as RSVP.
1. Find a route that supports resource reservation.
This is simply "type-of-service" routing, a facility that is
already available in some modern routing protocols.
2. Find a route that has sufficient unreserved capacity for a
new flow.
Early experiments on the ARPANET showed that it is difficult
to do load-dependent dynamic routing on a packet-by-packet
basis without instability problems. However, instability
should not be a problem if load-dependent routing is
performed only at reservation setup time.
Two different approaches might be taken to finding a route
with enough capacity. One could modify the routing
protocol(s) and interface them to the traffic control
mechanism, so the route computation can consider the average
recent load. Alternatively, the routing protocol could be
(re-)designed to provide multiple alternative routes, and
reservation setup could be attempted along each in turn.
3. Adapt to a route failure
When some node or link fails, adaptive routing finds an
alternate path. The periodic refresh messages of RSVP will
automatically request a reservation along the new path. Of
course, this reservation may fail because there is
insufficienct available capacity on the new path. This is a
problem of provisioning and network engineering, which cannot
be solved by the routing or setup protocols.
There is a problem of timeliness of establishing reservation
state on the new path. The end-to-end robustness mechanism
of refreshes is limited in frequency by overhead, which may
cause a gap in realtime service when an old route breaks and
a new one is chosen. It should be possible to engineer RSVP
to sypplement the global refresh mechanism with a local
repair mechanism, using hints about route changes from the
routing mechanism.
4. Adapt to a route change (without failure)
Route changes may occur even without failure in the affected
path. Although RSVP could use the same repair techniques as
those described in (3), this case raises a problem with the
robustness of the QoS guarantees. If it should happen that
admission control fails on the new route, the user will see
service degradation unnecessarily and capriciously, since the
orginal route is still functional.
To avoid this problem, a mechanism called "route pinning" has
been suggested. This would modify the routing protocol
implementation and the interface to the classifier, so that
routes associated with resource reservations would be
"pinned". The routing prootocol would not change a pinned
route if it was still viable.
It may eventually be possible to fold together the routing and
reservation setup problems, but we do not yet understand enough to
do that. Furthermore, the reservation protocol needs to coexist
with a number of different routing protocols in use in the
Internet. Therefore, RSVP is currently designed to work with any
current-generation routing protocol without modification. This is
a short-term compromise, which may result in an occasional failure
to create the best, or even any, real-time session, or an
occasional service degradation due to a route change. We expect
that future generations of routing protocols will remove this
compromise, by including hooks and mechanisms that, in conjunction
with RSVP, will solve the problems (1) through (4) just listed.
They will support route pinning, notification of RSVP to trigger
local repair, and selection of routes with "IS" support and
adequate capacity.
The last routing-related issue is provided by mobile hosts. Our
conjecture is that mobility is not essentially different from
other route changes, so that the mechanism suggested in (3) and
(4) will suffice. More study and experimentation is needed to
prove or disprove this conjecture.
6. ACKNOWLEDGMENTS
Many Internet researchers have contributed to the work described in
this memo. We want to especially acknowledge, Steve Casner, Steve
Deering, Deborah Estrin, Sally Floyd, Shai Herzog, Van Jacobson,
Sugih Jamin, Craig Partridge, John Wroclawski, and Lixia Zhang. This
approach to Internet integrated services was initially discussed and
organized in the End-to-End Research Group of the Internet Research
Taskforce, and we are grateful to all members of that group for their
interesting (and sometimes heated) discussions.
REFERENCES
[CerfKahn74] Cerf, V., and R. Kahn, "A Protocol for Packet Network
Intercommunication", IEEE Trans on Comm., Vol. Com-22, No. 5, May
1974.
[Clark88] Clark, D., "The Design Philosophy of the DARPA Internet
Protocols", ACM SIGCOMM '88, August 1988.
[CSZ92] Clark, D., Shenker, S., and L. Zhang, "Supporting Real-Time
Applications in an Integrated Services Packet Network: Architecture
and Mechanisms", Proc. SIGCOMM '92, Baltimore, MD, August 1992.
[DKS89] Demers, A., Keshav, S., and S. Shenker. "Analysis and
Simulation of a Fair Queueing Algorithm", Journal of
Internetworking: Research and Experience, 1, pp. 3-26, 1990. Also
in Proc. ACM SIGCOMM '89, pp 3-12.
[SCZ93a] Shenker, S., Clark, D., and L. Zhang, "A Scheduling Service
Model and a Scheduling Architecture for an Integrated Services
Packet Network", submitted to ACM/IEEE Trans. on Networking.
[SCZ93b] Shenker, S., Clark, D., and L. Zhang, "A Service Model for the
Integrated Services Internet", Work in Progress, October 1993.
[Floyd92] Floyd, S., "Issues in Flexible Resource Management for
Datagram Networks", Proceedings of the 3rd Workshop on Very High
Speed Networks, March 1992.
[Jacobson91] Jacobson, V., "Private Communication", 1991.
[JCSZ92] Jamin, S., Shenker, S., Zhang, L., and D. Clark, "An Admission
Control Algorithm for Predictive Real-Time Service", Extended
abstract, in Proc. Third International Workshop on Network and
Operating System Support for Digital Audio and Video, San Diego, CA,
Nov. 1992, pp. 73-91.
[Parekh92] Parekh, A., "A Generalized Processor Sharing Approach to
Flow Control in Integrated Services Networks", Technical Report
LIDS-TR-2089, Laboratory for Information and Decision Systems,
Massachusetts Institute of Technology, 1992.
[Partridge92] Partridge, C., "A Proposed Flow Specification", RFC 1363,
BBN, July 1992.
[RSVP93a] Zhang, L., Deering, S., Estrin, D., Shenker, S., and D.
Zappala, "RSVP: A New Resource ReSerVation Protocol", Accepted for
publication in IEEE Network, 1993.
[RSVP93b] Zhang, L., Braden, R., Estrin, D., Herzog, S., and S. Jamin,
"Resource ReSerVation Protocol (RSVP) - Version 1 Functional
Specification", Work in Progress, 1993.
[ST2-90] Topolcic, C., "Experimental Internet Stream Protocol: Version
2 (ST-II)", RFC 1190, BBN, October 1990.
[Tenet90] Ferrari, D., and D. Verma, "A Scheme for Real-Time Channel
Establishment in Wide-Area Networks", IEEE JSAC, Vol. 8, No. 3, pp
368-379, April 1990.
Security Considerations
As noted in Section 2.1, the ability to reserve resources will create
a requirement for authentication, both of users requesting resource
guarantees and of packets that claim to have the right to use those
guarantees. These authentication issues are not otherwise addressed
in this memo, but are for further study.
Authors' Addresses
Bob Braden
USC Information Sciences Institute
4676 Admiralty Way
Marina del Rey, CA 90292
Phone: (310) 822-1511
EMail: Braden@ISI.EDU
David Clark
MIT Laboratory for Computer Science
545 Technology Square
Cambridge, MA 02139-1986
Phone: (617) 253-6003
EMail: ddc@LCS.MIT.EDU
Scott Shenker
Xerox Palo Alto Research Center
3333 Coyote Hill Road
Palo Alto, CA 94304
Phone: (415) 812-4840
EMail: Shenker@PARC.XEROX.COM

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